Add log to detect the out of order RTCP packets.

media/cast/net/cast_transport_sender_impl.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/cast/net/cast_transport_sender_impl.h"
 
 #include <stddef.h>
 #include <algorithm>
 #include <string>
 #include <utility>
@@ skipping 356 matching lines @@
   } else if (video_sender_ && video_sender_->ssrc() == ssrc) {
     dedup_info.resend_interval = video_rtcp_session_->current_round_trip_time();
 
     // Only use audio stream to dedup if there is one.
     if (audio_sender_) {
       dedup_info.last_byte_acked_for_audio = last_byte_acked_for_audio_;
     }
   }
 
   if (!cast_message.missing_frames_and_packets.empty()) {
+    VLOG(2) << "feedback_count: " << cast_message.feedback_count;
     // This call does two things.
     // 1. Specifies that retransmissions for packets not listed in the set are
     //    cancelled.
     // 2. Specifies a deduplication window. For video this would be the most
     //    recent RTT. For audio there is no deduplication.
     ResendPackets(ssrc, cast_message.missing_frames_and_packets, true,
                   dedup_info);
   }
 
   if (!cast_message.received_later_frames.empty()) {
@@ skipping 99 matching lines @@
                "calling CastTransportSenderImpl::SendRtcpFromRtpReceiver.";
     return;
   }
   pacer_.SendRtcpPacket(rtcp_builder_at_rtp_receiver_->local_ssrc(),
                         rtcp_builder_at_rtp_receiver_->Finish());
   rtcp_builder_at_rtp_receiver_.reset();
 }
 
 }  // namespace cast
 }  // namespace media